The present invention relates to the field of devices for improving the speech perception of hearing impaired subjects. Such devices include acoustic hearing aids, tactile aids, cochlear prostheses and brain stem implants. In particular the invention is concerned with optimising the intelligibility of speech delivered to a subject by means of a directionally discriminating device.
In general the effects of hearing impairment are characterised by the undesirable conditioning of a speech signal along the subject""s hearing chain so as to result in attenuation and often distortion of the signal. It has been found that a standard hearing aid which amplifies the ambient sound can compensate for hearing losses attributable to attenuation, however such systems are of little assistance in low signal-to-noise ratio conditions.
Therefore, while most hearing aids provide substantial relief to the hearing impaired in single speaker, low reverberation, environments they are less useful where several speech sources are present simultaneously with the one of interest to the subject or when used in a room exhibiting strong reverberation characteristics. This poor performance is because such conditions are more likely to result in a lower speech signal-to-noise ratio than is prevalent in a single speaker, low reverberation environment. The problem of aiding a hearing impaired subject in a noisy environment is not overcome merely by indiscriminately amplifying both the speech of interest as well as the background noise.
In order to address this problem directional hearing aids have been used. Such hearing aids are able to spatially discriminate between sound sources. These aids selectively amplify sound sources in a particular direction or xe2x80x9cbeamxe2x80x9d relative to the aid.
A common method for producing spatial discrimination in a sound field has been to process the outputs from an array of microphones. Both fixed and adaptive array processors have been used. The principal property of adaptive array processors is that the microphone weights are continually adjusted with the array being statistically optimised according to some criterion. A problem with the adaptive array is that in a reverberant environment the processor may be unable to determine the direction of the desired signal and hence the weights to be adjusted. Consequently adaptive array processors will not be discussed further. In contrast, in a fixed array processor the signal processing components of the array are time-invariant, the fixed array using data independent weights and delays applied to the microphone outputs to create a maximum sensitivity to signals coming from a desired direction. There are many different types and configurations of fixed array processors. Each such processor has associated with it a degree of directivity and a particular level of inherent noise.
For example, one fixed processing arrangement which is well known in the art is based on addition and appropriate delay of the separate microphone outputs. Such an additive processor exhibits only moderate directionality however it can be used in relatively quiet environments because it has a greater signal-to-noise ratio than many more directional types of processor or indeed even a single microphone.
An alternative method for achieving spatial discrimination is to subtract the output of some of the separate microphones from the others. In this case the level of subtraction involved determines the amount by which off-beam sounds are suppressed. For example, second order subtraction, by which difference signals are subtracted from each other, affords a greater degree of suppression than first order subtraction, by which difference signals are added to each other, but has the disadvantage that whilst it strongly attenuates off-beam sounds it also attenuates the wanted signal to such an extent that internal microphone noise becomes significant when used in quiet surroundings. While subtractive processing has a higher directional performance than additive processing its ratio of signal to internal noise is poorer than that of the additive processor because of the increased on-beam attenuation. Details of the construction and theory of additive and subtractive sound processors are described in Speech Intelligibility Enhancement Technique Multi Microphone Array by G. Raicevich a Thesis for the degree of Master of Engineering, available from the library of the University of Technology Sydney, Broadway, Sydney, Australia.
In general, fixed array processors which have a relatively high directional performance and a relatively lower ratio of signal to internal noise, in quiet environments, such as the above described subtractive processor, are more suited to use in high noise situations. However in lower noise environments the converse is true and so it is preferable to use a fixed array which, whilst it may exhibit a lower directionality has the advantage of a significantly higher ratio of signal to internal noise.
Another type of fixed array processor is the so-called xe2x80x9cSupergainxe2x80x9d. The constrained supergain array described in the paper Practical Supergain, by Henry Cox et al (IEEE Transactions on Acoustics Speech and Signal Processing Vol ASSP 34 No. 3 June 1986), applies complex weights to the individual microphone outputs of a microphone array. By modifying a frontal gain constraint the values of the weights may be calculated and the qualities of the supergain array may be controlled. For example, a supergain array processor may incorporate weighting which takes the background noise characteristics into account. In high background noise the resulting processor will have a lower frontal gain constraint and hence a higher directivity. The associated higher array internal noise will not unduly affect the signal-to-noise ratio. Conversely, when designing a supergain processor for use in a low background noise level environment, a higher frontal gain constraint will result in a lower directivity but a lower array internal noise.
While a fixed array processor may operate well in a particular noise environment its characteristics may not be ideal for operation as the background noise level of its environment changes. Consequently there is a need for a hearing aid which, while compact is also directional and able to adjust to changes in the noise environment without the drawbacks associated with adaptive arrays.
It is an object of the present invention to provide a system by which a fixed array processing strategy is determined according to prevailing environmental conditions, preferably the level of ambient noise floor, so as to maximise the speech signal-to-noise ratio of a spatially discriminating aid for the hearing impaired.
It is a further object of the present invention to provide a system by which the complex weights of a constrained supergain array may be adjusted in order to maximise the overall signal-to-noise ratio of the processor given the prevailing acoustic environment in which the array is used.
According to a first aspect of the present invention there is provided a method for processing sound comprising the steps of:
a) determining the signal-to-noise performance of a plurality of fixed microphone array processors for a range of ambient noise levels;
b) monitoring a parameter indicative of ambient noise conditions to determine the prevailing ambient noise level;
c) determining the operating parameters of a microphone array processor being the microphone array processor of said plurality of microphone array proccessors having the highest signal-to-noise performance in the prevailing ambient noise level; and
d) processing the output of a microphone array with a processor having the operating parameters of the processor selected in step c).
According to a further aspect of the present invention there is provided An apparatus for processing sound comprising:
a) an array of microphones;
b) first and second array processors coupled to said array, each of said processors arranged to produce a characteristic total noise output being a function of ambient noise floor,
the first processor being arranged to produce a lower characteristic total noise output than the second processor over a first range of values of noise floor, and the second processor being arranged to produce a lower characteristic total noise output than the first processor over a second range of values of said noise floor;
c) a noise floor indicating circuit coupled to at least one microphone of said array of microphones arranged to produce a noise floor signal indicative of said ambient noise floor;
d) control means coupled to said noise floor indicating circuit and arranged to produce first and second control signals indicating when said noise floor signal is in said first range of values or in said second range of values;
e) first and second variable gain means,
the first and second variable gain means being coupled to the first and second microphone array processor, and being responsive to the first and second control signal respectively,
the first and second variable gain means arranged to apply variable gain to the characteristic total noise output of the first and second array processor, respectively,
the control means and said first and second variable gain means being further arranged so that the gain applied by the first variable gain means is greater than the gain applied by the second variable gain means when said noise floor signal is within said first range and the gain applied by the second variable gain means is greater than the gain applied by the first variable gain means when said noise floor is within said second range.
According to a further aspect of the invention there is provided an apparatus for processing sound comprising:
a) an array of microphones;
b) first and second array processors coupled to said array of microphones, each of said processors arranged to produce a characteristic total noise output being a function of ambient noise,
the first processor being arranged to produce a lower characteristic total noise output than the second processor over a first range of values of ambient noise, and the second processor being arranged to produce a lower characteristic total noise output than the first processor over a second range of values of said ambient noise;
c) background noise processor coupled to said microphone array and arranged to have maximum sensitivity to background noise said background noise processor producing a background noise signal;
d) on-beam signal detect circuit responsive to said background noise processor and to output from said first and second array processors for producing a detect signal indicative of the presence of on-beam signal;
e) first and second sample-and-hold circuits coupled to said first and second processors respectively and being responsive to said detect signal, said first and second sample-and-hold circuits arranged to produce first and second ambient noise estimates;
f) control circuit coupled to said first and second sample-and-hold circuits and arranged to produce first and second control signals to indicate relative magnitudes of said first and second noise estimates;
g) first and second variable gain circuits coupled to the first and second array processor, respectively,
the first and second variable gain circuits being responsive to the first and second control signal, respectively,
and arranged to apply variable gain to the characteristic total noise output of the first and second array processor, respectively,
the control circuit and said first and second variable gain circuits being further arranged so that when said first noise estimate is less than said second noise estimate said gain applied by the first variable gain circuit is greater than said gain applied by the second variable gain circuit and when said second noise estimate is less than said first noise estimate said gain applied by the second variable gain circuit is greater than said gain applied by the first variable gain circuit.
According to a final aspect there is provided an apparatus for processing sound comprising:
a) a microphone array comprising a plurality of microphones each microphone producing a signal corresponding to surrounding ambient sound;
b) a plurality of antialiasing filters coupled to each microphone of said array respectively, each antialiasing filter arranged to produce a low-pass filtered signal;
c) a plurality of analog to digital converters coupled to each antialiasing filter respectively, each analog to digital converter arranged to produce a digital noise signal;
d) a microphone array processor coupled to said analog to digital converters arranged to produce a noise level signal indicative of the ambient noise level;
e) an allocation means responsive to said noise level signal and arranged to produce a plurality of weighting signals in accordance with a predetermined rule;
f) a plurality of digital multiplier means coupled to each analog to digital converter and respectively responsive to said plurality of weighting signals, each said multiplier arranged to perform a complex multiplication operation on each digital noise signal respectively, said plurality of digital multipliers producing a corresponding plurality of multiplied signals;
g) means for delivering said plurality of multiplied signals for further processing.
The invention also extends to acoustic hearing aids, tactile aids, cochlear prostheses, brain stem implants and other aids to hearing which incorporate the above described inventive features.